Benny Prijono wrote: > On Thu, Aug 14, 2008 at 9:47 AM, Pedro Gon?alves > <pedro.pandre at gmail.com <mailto:pedro.pandre at gmail.com>> wrote: > > I tried again and this time I see that RTCP is going from the > caller to > the answerer. > The problem is that RTP is not. > Any possible ideas why this is happening? > > > Check the audio troubleshooting Wiki: > http://trac.pjsip.org/repos/wiki/sound-problems I already did that :( However, those instructions are for use with PJSUA, and as you know I am using PJSIP at a lower level (PJNATH / PJLIB / etc.) I tried to translate those instructions to what was happening in the code, and everything looks normal. As no RTP packets are going from caller to answerer, I think the best troubleshooting guide for me is http://trac.pjsip.org/repos/wiki/audio-problem-remote-no-audio 1. Check that correct audio device is being used <http://trac.pjsip.org/repos/wiki/audio-check-correct-device> The correct audio device is being used, because my application has a configuration setting which allows us to select the audio device, to adjust volume and to hear from it. I can hear myself perfectly. 1. Check that microphone is functioning properly (level is correct etc.) by looping the microphone to the speaker with pjsua <http://trac.pjsip.org/repos/wiki/audio-check-loopback> Same as above 1. Check that no other application is using the sound devices. It is common to not be able to use sound device when other application is using the device. No other app is using the device. Only my app is running 1. Check that the call's media is connected to the sound device in the conference bridge <http://trac.pjsip.org/repos/wiki/audio-check-conf-connection> I think this is being done correctly: 19:36:20.500 strm1631D8AC VAD temporarily disabled 19:36:20.500 rtp.c pjmedia_rtp_session_init: ses=1631EAD4, default_pt=0, ssrc=0x4ae13d6c 19:36:20.562 rtp.c pjmedia_rtp_session_init: ses=1631F6DC, default_pt=0, ssrc=0x4ae13d6c 19:36:20.578 stream.c Stream strm1631D8AC created 19:36:20.578 strm1631D8AC Encoder stream started 19:36:20.578 strm1631D8AC Decoder stream started 19:36:20.703 pasound.c Opened device Realtek AC97 Audio(MME)/Realtek AC97 Audio(MME) for recording and playback, sample rate=8000, ch=1, bits=16, 400 samples per frame, input latency=100 ms, output latency=100 ms 19:36:20.703 pasound.c Starting Realtek AC97 Audio stream.. 19:36:20.703 pasound.c PA message: Pa_StartStream: waveInStart returned = 0x0. 19:36:20.703 pasound.c Done, status=0 19:36:20.703 ec16662EF8 Creating AEC 19:36:20.750 pasound.c Player thread started 19:36:20.750 pasound.c Recorder thread started 19:36:20.921 ec16662EF8 AEC created, clock_rate=8000, channel=1, samples per frame=400, tail length=800 ms, latency=100 ms 19:36:20.921 conference.c Port 1 (strm1631D8AC) transmitting to port 0 (Master/sound) 19:36:20.921 conference.c Port 0 (Master/sound) transmitting to port 1 (strm1631D8AC) (you can see the complete log and capture in http://student.dei.uc.pt/~pandre/caller_not_sending_media2.zip) 1. Check that pjsua is transmitting RTP packet to the correct address <http://trac.pjsip.org/repos/wiki/audio-check-remote-address> This is where the problem starts. No RTP packets are leaving the caller, and I don't have any idea why. Strangely, RTCP packets are going to the correct address. 1. Use pjsua on the remote side to check that packets are received. Follow the instruction on this page <http://trac.pjsip.org/repos/wiki/audio-check-rx-rtp> on how to use pjsua to verify receipt of incoming RTP packets. 2. Check that CPU utilization is not too high <http://trac.pjsip.org/repos/wiki/audio-check-cpu> 3. Check that echo canceller doesn't cut audio too aggressively? <http://trac.pjsip.org/repos/wiki/audio-check-ec> When using ICE, wasn't PJSIP supposed to send RTP/RTCP packets between the initial answer and the subsequent offer, to the address negotiated in the initial offer/answer? Best Regards Pedro Gon?alves
